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r219139 | mnicholson | 2009-09-17 10:18:01 -0500 (Thu, 17 Sep 2009) | 17 lines
Merged revisions 219136 via svnmerge from
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r219136 | mnicholson | 2009-09-17 09:58:39 -0500 (Thu, 17 Sep 2009) | 10 lines
Prevent a potential race condition and crash when hanging up a channel by removing the channel from the channel list before begining channel tear down.
This fix may potentially cause problems with CDR backends that access the channel a CDR is associated with via the channel list. This fix makes the channel unavabile at the time when the CDR backend is invoked. This has been documented in include/asterisk/cdr.h.
(closes issue #15316)
Reported by: vmarrone
Tested by: mnicholson
Review: https://reviewboard.asterisk.org/r/362/
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r201056 | kpfleming | 2009-06-16 13:54:30 -0500 (Tue, 16 Jun 2009) | 18 lines
Merged revisions 200991 via svnmerge from
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r200991 | kpfleming | 2009-06-16 12:05:38 -0500 (Tue, 16 Jun 2009) | 11 lines
Improve support for media paths that can generate multiple frames at once.
There are various media paths in Asterisk (codec translators and UDPTL, primarily)
that can generate more than one frame to be generated when the application calling
them expects only a single frame. This patch addresses a number of those cases,
at least the primary ones to solve the known problems. In addition it removes the
broken TRACE_FRAMES support, fixes a number of bugs in various frame-related API
functions, and cleans up various code paths affected by these changes.
https://reviewboard.asterisk.org/r/175/
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r198856 | dvossel | 2009-06-02 16:17:49 -0500 (Tue, 02 Jun 2009) | 10 lines
Generic call forward api, ast_call_forward()
The function ast_call_forward() forwards a call to an extension specified in an ast_channel's call_forward string. After an ast_channel is called, if the channel's call_forward string is set this function can be used to forward the call to a new channel and terminate the original one. I have included this api call in both channel.c's ast_request_and_dial() and feature.c's feature_request_and_dial(). App_dial and app_queue already contain call forward logic specific for their application and options.
(closes issue #13630)
Reported by: festr
Review: https://reviewboard.asterisk.org/r/271/
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The bridge was terminating immediately after the attended transfer was
completed. The problem was because upon reentering ast_channel_bridge
nexteventts was checked to see if it was set and if so could possibly
return AST_BRIDGE_COMPLETE.
(closes issue #15183)
Reported by: andrebarbosa
Tested by: andrebarbosa, tootai, loloski
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r192318 | kpfleming | 2009-05-05 12:34:19 +0200 (Tue, 05 May 2009) | 5 lines
Properly account for memory allocated for channels and datastores
As in previous commits, when channels are allocated (with ast_channel_alloc) or datastores are allocated (with ast_datastore_alloc) properly account for the memory being owned by the caller, instead of the allocator function itself.
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r190057 | jpeeler | 2009-04-22 16:15:55 -0500 (Wed, 22 Apr 2009) | 9 lines
Fix building of chan_h323 with gcc-3.3
There seems to be a bug with old versions of g++ that doesn't allow a structure
member to use the name list. Rename list member to group_list in ast_group_info
and change the few places it is used.
(closes issue #14790)
Reported by: stuarth
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r184762 | kpfleming | 2009-03-27 14:10:32 -0500 (Fri, 27 Mar 2009) | 12 lines
Improve timing interface to remember which provider provided a timer
The ability to load/unload timing interfaces is nice, but it means that when a timer is allocated, it may come from provider A, but later provider B becomes the 'preferred' provider. If this happens, all timer API calls on the timer that was provided by provider A will actually be handed to provider B, which will say WTF and return an error.
This patch changes the timer API to include a pointer to the provider of the timer handle so that future operations on the timer will be forwarded to the proper provider.
(closes issue #14697)
Reported by: moy
Review: http://reviewboard.digium.com/r/211/
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r182847 | russell | 2009-03-17 21:28:55 -0500 (Tue, 17 Mar 2009) | 52 lines
Merged revisions 182810 via svnmerge from
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r182810 | russell | 2009-03-17 21:09:13 -0500 (Tue, 17 Mar 2009) | 44 lines
Fix cases where the internal poll() was not being used when it needed to be.
We have seen a number of problems caused by poll() not working properly on
Mac OSX. If you search around, you'll find a number of references to using
select() instead of poll() to work around these issues. In Asterisk, we've
had poll.c which implements poll() using select() internally. However, we
were still getting reports of problems.
vadim investigated a bit and realized that at least on his system, even
though we were compiling in poll.o, the system poll() was still being used.
So, the primary purpose of this patch is to ensure that we're using the
internal poll() when we want it to be used.
The changes are:
1) Remove logic for when internal poll should be used from the Makefile.
Instead, put it in the configure script. The logic in the configure
script is the same as it was in the Makefile. Ideally, we would have
a functionality test for the problem, but that's not actually possible,
since we would have to be able to run an application on the _target_
system to test poll() behavior.
2) Always include poll.o in the build, but it will be empty if AST_POLL_COMPAT
is not defined.
3) Change uses of poll() throughout the source tree to ast_poll(). I feel
that it is good practice to give the API call a new name when we are
changing its behavior and not using the system version directly in all cases.
So, normally, ast_poll() is just redefined to poll(). On systems where
AST_POLL_COMPAT is defined, ast_poll() is redefined to ast_internal_poll().
4) Change poll() in main/poll.c to be ast_internal_poll().
It's worth noting that any code that still uses poll() directly will work fine
(if they worked fine before). So, for example, out of tree modules that are
using poll() will not stop working or anything. However, for modules to work
properly on Mac OSX, ast_poll() needs to be used.
(closes issue #13404)
Reported by: agalbraith
Tested by: russell, vadim
http://reviewboard.digium.com/r/198/
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r182525 | kpfleming | 2009-03-17 09:38:11 -0500 (Tue, 17 Mar 2009) | 11 lines
Improve behavior of ast_answer() to not lose incoming frames
ast_answer(), when supplied a delay before returning to the caller, use ast_safe_sleep() to implement the delay. Unfortunately during this time any incoming frames are discarded, which is problematic for T.38 re-INVITES and other sorts of channel operations.
When a delay is not passed to ast_answer(), it still delays for up to 500 milliseconds, waiting for media to arrive. Again, though, it discards any control frames, or non-voice media frames.
This patch rectifies this situation, by storing all incoming frames during the delay period on a list, and then requeuing them onto the channel before returning to the caller.
http://reviewboard.digium.com/r/196/
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r180719 | jpeeler | 2009-03-09 15:58:17 -0500 (Mon, 09 Mar 2009) | 16 lines
Add Doxygen documentation for API changes from 1.6.0 to 1.6.1
Copied from my review board description:
This is a continuation of the API changes documentation started for describing
changes between releases. Most of the API changes were pretty simple needing
only to be brought to attention via the new "Asterisk API Changes" list.
However, if you see anything that needs further explanation feel free to
supplement what is there. The current method of documenting is to add (in the
header file): \version <ver number> <description of changes> and then to add
the function to the change list in doxyref.h on the AstAPIChanges page. I also
made sure all the functions that were newly added were tagged with \since
1.6.1. I think this is a good habit to start both for the historical aspect as
well as for the future ability to easily add a "New Asterisk API" page.
Review: http://reviewboard.digium.com/r/190/
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r177387 | jpeeler | 2009-02-19 10:45:02 -0600 (Thu, 19 Feb 2009) | 3 lines
Fix another merge error from 176708
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r176708 | jpeeler | 2009-02-17 16:08:00 -0600 (Tue, 17 Feb 2009) | 23 lines
Merged revisions 176701 via svnmerge from
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r176701 | jpeeler | 2009-02-17 15:54:34 -0600 (Tue, 17 Feb 2009) | 17 lines
Modify bridging to properly evaluate DTMF after first warning is played
The main problem is currently if the Dial flag L is used with a warning sound,
DTMF is not evaluated after the first warning sound. To fix this, a flag has
been added in ast_generic_bridge for playing the warning which ensures that if
a scheduled warning is missed, multiple warrnings are not played back (due to a
feature evaluation or waiting for digits). ast_channel_bridge was modified to
store the nexteventts in the ast_bridge_config structure as that information
was lost every time ast_channel_bridge was reentered, causing a hangup due to
incorrect time calculations.
(closes issue #14315)
Reported by: tim_ringenbach
Reviewed on reviewboard:
http://reviewboard.digium.com/r/163/
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r174945 | mmichelson | 2009-02-11 16:41:01 -0600 (Wed, 11 Feb 2009) | 29 lines
Fix 'd' option for app_dial and add new option to Answer application
The 'd' option would not work for channel types which use RTP to transport
DTMF digits. The only way to allow for this to work was to answer the channel
if we saw that this option was enabled.
I realized that this may cause issues with CDRs, specifically with giving false
dispositions and answer times. I therefore modified ast_answer to take another
parameter which would tell if the CDR should be marked answered.
I also extended this to the Answer application so that the channel may be answered
but not CDRified if desired.
I also modified app_dictate and app_waitforsilence to only answer the channel if it
is not already up, to help not allow for faulty CDR answer times.
All of these changes are going into Asterisk trunk. For 1.6.0 and 1.6.1, however, all
the changes except for the change to the Answer application will go in since we do
not introduce new features into stable branches
(closes issue #14164)
Reported by: DennisD
Patches:
14164.patch uploaded by putnopvut (license 60)
Tested by: putnopvut
Review: http://reviewboard.digium.com/r/145
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r172598 | mmichelson | 2009-01-30 16:22:04 -0600 (Fri, 30 Jan 2009) | 3 lines
Fix redefinition of flag in channel.h
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r172063 | murf | 2009-01-28 13:31:06 -0700 (Wed, 28 Jan 2009) | 52 lines
Merged revisions 172030 via svnmerge from
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r172030 | murf | 2009-01-28 11:51:16 -0700 (Wed, 28 Jan 2009) | 46 lines
This patch fixes h-exten running misbehavior in manager-redirected
situations.
What it does:
1. A new Flag value is defined in include/asterisk/channel.h,
AST_FLAG_BRIDGE_HANGUP_DONT, which used as a messenge to the
bridge hangup exten code not to run the h-exten there (nor
publish the bridge cdr there). It will done at the pbx-loop
level instead.
2. In the manager Redirect code, I set this flag on the channel
if the channel has a non-null pbx pointer. I did the same for the
second (chan2) channel, which gets run if name2 is set...
and the first succeeds.
3. I restored the ending of the cdr for the pbx loop h-exten
running code. Don't know why it was removed in the first place.
4. The first attempt at the fix for this bug was to place code
directly in the async_goto routine, which was called from a
large number of places, and could affect a large number of
cases, so I tested that fix against a fair number of transfer
scenarios, both with and without the patch. In the process,
I saw that putting the fix in async_goto seemed not to affect
any of the blind or attended scenarios, but still, I was
was highly concerned that some other scenarios I had not tested
might be negatively impacted, so I refined the patch to
its current scope, and jmls tested both. In the process, tho,
I saw that blind xfers in one situation, when the one-touch
blind-xfer feature is used by the peer, we got strange
h-exten behavior. So, I inserted code to swap CDRs and
to set the HANGUP_DONT field, to get uniform behavior.
5. I added code to the bridge to obey the HANGUP_DONT flag,
skipping both publishing the bridge CDR, and running
the h-exten; they will be done at the pbx-loop (higher)
level instead.
6. I removed all the debug logs from the patch before committing.
7. I moved the AUTOLOOP set/reset in the h-exten code in res_features
so it's only done if the h-exten is going to be run. A very
minor performance improvement, but technically correct.
(closes issue #14241)
Reported by: jmls
Patches:
14241_redirect_no_bridgeCDR_or_h_exten_via_transfer uploaded by murf (license 17)
Tested by: murf, jmls
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r168562 | russell | 2009-01-13 13:22:13 -0600 (Tue, 13 Jan 2009) | 10 lines
Merged revisions 168561 via svnmerge from
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r168561 | russell | 2009-01-13 13:13:05 -0600 (Tue, 13 Jan 2009) | 2 lines
Revert unnecessary indications API change from rev 122314
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r164419 | mmichelson | 2008-12-15 13:51:24 -0600 (Mon, 15 Dec 2008) | 12 lines
Merged revisions 164416 via svnmerge from
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r164416 | mmichelson | 2008-12-15 13:45:07 -0600 (Mon, 15 Dec 2008) | 4 lines
Add notes to autoservice and pbx doxygen regarding a potential
deadlock scenario so that it is avoided in the future
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r163449 | russell | 2008-12-12 07:55:30 -0600 (Fri, 12 Dec 2008) | 34 lines
Merged revisions 163448 via svnmerge from
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r163448 | russell | 2008-12-12 07:44:08 -0600 (Fri, 12 Dec 2008) | 26 lines
Resolve issues that could cause DTMF to be processed out of order.
These changes come from team/russell/issue_12658
1) Change autoservice to put digits on the head of the channel's frame readq
instead of the tail. If there were frames on the readq that autoservice
had not yet read, the previous code would have resulted in out of order
processing. This required a new API call to queue a frame to the head
of the queue instead of the tail.
2) Change up the processing of DTMF in ast_read(). Some of the problems
were the result of having two sources of pending DTMF frames. There
was the dtmfq and the more generic readq. Both were used for pending
DTMF in various scenarios. Simplifying things to only use the frame
readq avoids some of the problems.
3) Fix a bug where a DTMF END frame could get passed through when it
shouldn't have. If code set END_DTMF_ONLY in the middle of digit emulation,
and a digit arrived before emulation was complete, digits would get
processed out of order.
(closes issue #12658)
Reported by: dimas
Tested by: russell, file
Review: http://reviewboard.digium.com/r/85/
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r159818 | kpfleming | 2008-11-29 11:57:39 -0600 (Sat, 29 Nov 2008) | 18 lines
incorporates r159808 from branches/1.4:
------------------------------------------------------------------------
r159808 | kpfleming | 2008-11-29 10:58:29 -0600 (Sat, 29 Nov 2008) | 7 lines
update dev-mode compiler flags to match the ones used by default on Ubuntu Intrepid, so all developers will see the same warnings and errors
since this branch already had some printf format attributes, enable checking for them and tag functions that didn't have them
format attributes in a consistent way
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in addition:
move some format attributes from main/utils.c to the header files they belong in, and fix up references to the relevant functions based on new compiler warnings
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r157306 | mmichelson | 2008-11-18 12:31:08 -0600 (Tue, 18 Nov 2008) | 20 lines
Merged revisions 157305 via svnmerge from
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r157305 | mmichelson | 2008-11-18 12:25:55 -0600 (Tue, 18 Nov 2008) | 12 lines
Fix a crash in the end_bridge_callback of app_dial and
app_followme which would occur at the end of an attended
transfer. The error occurred because we initially stored
a pointer to an ast_channel which then was hung up due
to a masquerade.
This commit adds a "fixup" callback to the bridge_config
structure to allow for end_bridge_callback_data to be
changed in the case that a new channel pointer is needed
for the end_bridge_callback.
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r155554 | seanbright | 2008-11-08 20:27:00 -0500 (Sat, 08 Nov 2008) | 14 lines
Merged revisions 155553 via svnmerge from
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r155553 | seanbright | 2008-11-08 20:08:07 -0500 (Sat, 08 Nov 2008) | 6 lines
Use static functions here instead of nested ones. This requires a small
change to the ast_bridge_config struct as well. To understand the reason
for this change, see the following post:
http://gcc.gnu.org/ml/gcc-help/2008-11/msg00049.html
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r153181 | twilson | 2008-10-31 13:55:33 -0500 (Fri, 31 Oct 2008) | 5 lines
Recent CDR fixes moved execution of the 'h' exten into the bridging code, so variables that were set after ast_bridge_call was called would not show up in the 'h' exten. Added a callback function to handle setting variables, etc. from w/in the bridging code. Calls back into a nested function within the function calling ast_bridge_call
(closes issue #13793)
Reported by: greenfieldtech
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r142676 | murf | 2008-09-11 22:50:48 -0600 (Thu, 11 Sep 2008) | 40 lines
Merged revisions 142675 via svnmerge from
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r142675 | murf | 2008-09-11 22:29:34 -0600 (Thu, 11 Sep 2008) | 29 lines
Tested by: sergee, murf, chris-mac, andrew, KNK
This is a "second attempt" to restore the previous "endbeforeh" behavior
in 1.4 and up. In order to capture information concerning all the
legs of transfers in all their infinite combinations, I was forced
to this particular solution by a chain of logical necessities, the
first being that I was not allowed to rewrite the CDR mechanism from
the ground up!
This change basically leaves the original machinery alone, which allows
IVR and local channel type situations to generate CDR's as normal, but
a channel flag can be set to suppress the normal running of the h exten.
That flag would be set by the code that runs the h exten from the
ast_bridge_call routine, to prevent the h exten from being run twice.
Also, a flag in the ast_bridge_config struct passed into ast_bridge_call
can be used to suppress the running of the h exten in that routine. This
would happen, for instance, if you use the 'g' option in the Dial app.
Running this routine 'early' allows not only the CDR() func to be used
in the h extension for reading CDR variables, but also allows them to
be modified before the CDR is posted to the backends.
While I dearly hope that this patch overcomes all problems, and
introduces no new problems, reality suggests that surely someone
will have problems. In this case, please re-open 13251 (or 13289),
and we'll see if we can't fix any remaining issues.
** trunk note: some code to suppress the h exten being run
from app_queue was added; for the 'continue' option available
only in trunk/1.6.x.
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in voicemail.
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channel related, and add the ability to add/find/remove datastores to manager sessions
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@129307 f38db490-d61c-443f-a65b-d21fe96a405b
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AMI commands can display that a channel is under control of an AGI.
Work inspired by work at customer site, but paid for by Edvina AB
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@128240 f38db490-d61c-443f-a65b-d21fe96a405b
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'dialcontext' is only going to be set once during the channel's lifetime, so make it a string field instead of a char array
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@126960 f38db490-d61c-443f-a65b-d21fe96a405b
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alignment requirements, saving 36 bytes per instance on a 64-bit platform
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@126187 f38db490-d61c-443f-a65b-d21fe96a405b
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- Convert chan_iax2 to use the timing API
- Convert usage of timing in the core to use the timing API instead of
using DAHDI directly
- Make a change to the timing API to add the set_rate() function
- change the timing core to use a rwlock
- merge a timing implementation, res_timing_dahdi
Basic testing was successful using res_timing_dahdi
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@122523 f38db490-d61c-443f-a65b-d21fe96a405b
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Configuration file and dialplan backwards compatability has been put in place where appropiate. Release announcement to follow.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@122234 f38db490-d61c-443f-a65b-d21fe96a405b
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- make data member of the ast_frame struct a named union instead of a void
Recently the ast_queue_hangup function got a new parameter, the hangupcause
Feedback came in that this is no good and that instead a new function should be created.
This I did.
The hangupcause was stored in the seqno member of the ast_frame struct. This is not very
elegant, and since there's already a data member that one should be used.
Problem is, this member was a void *.
Now it's a named union so it can hold a pointer, an uint32 and there's a padding in case someone
wants to store another type in there in the future.
This commit is so massive, because all ast_frame.data uses have to be
altered to ast_frame.data.data
Thanks russellb and kpfleming for the feedback.
(closes issue #12674)
Reported by: mvanbaak
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@117802 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r116463 | russell | 2008-05-14 16:32:00 -0500 (Wed, 14 May 2008) | 4 lines
Add ast_assert(), which can be used to handle fatal errors. It is only compiled
in if dev-mode is enabled, and only aborts if DO_CRASH is defined.
(inspired by issue #12650)
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@116469 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #10540)
Reported by: spendergrass
Patches:
20080417__bug10540.diff.txt uploaded by Corydon76 (license 14)
Tested by: blitzrage
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@115076 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #11328)
Reported by: rain
Patches:
20071207__pass_cause_in_hangup_control_frame.diff.txt uploaded by Corydon76 (license 14)
brought up-to-date to trunk by me
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@114637 f38db490-d61c-443f-a65b-d21fe96a405b
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Reported by: oej
Tested by: jpeeler
This patch implements multiple parking lots for parked calls. The default parkinglot is used by default, however setting the channel variable PARKINGLOT in the dialplan will allow use of any other configured parkinglot. See configs/features.conf.sample for more details on setting up another non-default parkinglot. Also, one can (currently) set the default parkinglot to use in the driver configuration file via the parkinglot option.
Patch initially written by oej, brought up to date and finalized by mvanbaak, and then stabilized and converted to astobj2 by me.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@114487 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r108583 | russell | 2008-03-13 16:38:16 -0500 (Thu, 13 Mar 2008) | 11 lines
Fix another issue that was causing crashes in chanspy. This introduces a new
datastore callback, called chan_fixup(). The concept is exactly like the
fixup callback that is used in the channel technology interface. This callback
gets called when the owning channel changes due to a masquerade. Before this
was introduced, if a masquerade happened on a channel being spyed on, the
channel pointer in the datastore became invalid.
(closes issue #12187)
(reported by, and lots of testing from atis)
(props to file for the help with ideas)
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@108584 f38db490-d61c-443f-a65b-d21fe96a405b
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channel. This uses control frames (so they can even pass across IAX2) to negotiate T38 and provided a way of getting the current status of T38 using queryoption. This should by no means cause any issues and if it does I will take responsibility for it.
(closes issue #11873)
Reported by: dimas
Patches:
v4-t38-api.patch uploaded by dimas (license 88)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@103799 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #11268)
Reported by: moy
Patches:
chantrace-datastored-encapsulated-rev94934.patch uploaded by moy (license 222)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@103754 f38db490-d61c-443f-a65b-d21fe96a405b
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suggestion from (const char *) Kevin. Thanks!
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@101154 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r100581 | russell | 2008-01-28 11:15:41 -0600 (Mon, 28 Jan 2008) | 9 lines
Make some deadlock related fixes. These bugs were discovered and reported
internally at Digium by Steve Pitts.
- Fix up chan_local to ensure that the channel lock is held before the local
pvt lock.
- Don't hold the channel lock when executing the timing function, as it can
cause a deadlock when using chan_local. This actually changes the code back
to be how it was before the change for issue #10765. But, I added some other
locking that I think will prevent the problem reported there, as well.
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@100582 f38db490-d61c-443f-a65b-d21fe96a405b
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as a channel variable BRIDGEPVTCALLID
This is important for call tracing in log files and CDRs, so that
the SIP callID can be traced along servers.
The CHANNEL dialplan function won't work here, since the outbound
channel is gone when we need the Call-ID.
Other channel drivers may now implement the same function :-),
but this patch only supports chan_sip.so.
Inspired by (issue #11816)
Reported by: ctooley
Patch by oej
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@99644 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@96368 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@95411 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r90735 | mmichelson | 2007-12-03 17:12:17 -0600 (Mon, 03 Dec 2007) | 22 lines
A big one...
This is the merge of the forward-loop branch. The main change here is that call-forwards can no longer loop.
This is accomplished by creating a datastore on the calling channel which has a linked list of all devices
dialed. If a forward happens, then the local channel which is created inherits the datastore. If, through this
progression of forwards and datastore inheritance, a device is attempted to be dialed a second time, it will simply
be skipped and a warning message will be printed to the CLI. After the dialing has been completed, the datastore
is detached from the channel and destroyed.
This change also introduces some side effects to the code which I shall enumerate here:
1. Datastore inheritance has been backported from trunk into 1.4
2. A large chunk of code has been removed from app_dial. This chunk is the section of code
which handles the call forward case after the channel has been requested but before it has
been called. This was removed because call-forwarding still works fine without it, it makes the
code less error-prone should it need changing, and it made this set of changes much less painful
to just have the forwarding handled in one place in each module.
3. Two new files, global_datastores.h and .c have been added. These are necessary since the datastore
which is attached to the channel may be created and attached in either app_dial or app_queue, so they
need a common place to find the datastore info. This approach was taken in case similar datastores are
needed in the future, there will be a common place to add them.
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@90873 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r90548 | file | 2007-12-03 14:40:56 -0400 (Mon, 03 Dec 2007) | 2 lines
Preserve the indication currently playing on a channel when a masquerade operation happens. (issue #BE-88)
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@90550 f38db490-d61c-443f-a65b-d21fe96a405b
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This works in much the same way as the automonitor, except that instead of using the monitor
app, it uses the mixmonitor app. By providing an 'x' or 'X' as a dial or queue option, a DTMF
sequence may be entered (as defined in features.conf) to start the one-touch mixmonitor.
This patch also introduces some new API calls to the audiohooks code for searching for an audiohook
by type and for searching for a running audiohook by type.
Big thanks to joetester for writing the initial patch, testing it and patiently waiting for it to
be committed.
(closes issue #10185, reported and patched by xmarksthespot)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@90388 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r90145 | russell | 2007-11-28 18:20:34 -0600 (Wed, 28 Nov 2007) | 5 lines
This set of changes is to make some callerID handling thread-safe.
The ast_set_callerid() function needed to lock the channel. Also, the handlers
for the CALLERID() dialplan function needed to lock the channel when reading
or writing callerid values directly on the channel structure.
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@90146 f38db490-d61c-443f-a65b-d21fe96a405b
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This was mostly to note whether a channel needed to be locked or not before
calling these functions. However, I added some other things, too.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@90139 f38db490-d61c-443f-a65b-d21fe96a405b
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